Legal claims defining the scope of protection. Each claim is shown in both the original legal language and a plain English translation.
1. A method, comprising: accessing, by a first network device comprising a processor, gap data related to a gap in voice packet data, wherein the gap is indicative of a lack of a voice activity and the gap is related to a handover condition associated with a mobile device and network devices other than the first network device, wherein the handover condition is related to a handover of a signal of the mobile device from being communicated from the first network device to being communicated from a second network device of the network devices; analyzing, by the first network device, the gap data to determine a punctuation type; based on the punctuation type, deferring, by the first network device, the handover of the signal, resulting in a deferred handover of the signal; based on the gap data, increasing, by the first network device, a memory size of a dejitter buffer, resulting in an increased memory size of the dejitter buffer; and analyzing, by the first network device, text data associated with the voice packet data to verify the punctuation type is represented in the text data.
This invention relates to improving voice communication quality during handover in mobile networks. The problem addressed is the disruption in voice calls when a mobile device transitions between network devices, often causing gaps in voice packet data due to handover delays. The solution involves a method where a first network device detects a gap in voice packet data, which indicates a lack of voice activity and is related to a handover condition. The network device analyzes the gap data to determine the type of punctuation (e.g., pause, comma, period) it represents. Based on the punctuation type, the handover of the mobile device's signal to a second network device is deferred to avoid interrupting critical speech segments. Additionally, the network device increases the memory size of a dejitter buffer to accommodate the delayed handover, ensuring smooth voice transmission. The system also verifies the punctuation type by analyzing text data associated with the voice packet data, ensuring accuracy in handover decisions. This approach minimizes call disruptions by intelligently timing handovers around natural speech pauses.
2. The method of claim 1 , further comprising: sending, by the first network device, data related to the handover condition being satisfied to the mobile device.
A method for managing network handover in wireless communication systems addresses the challenge of efficiently transferring a mobile device between network nodes to maintain connectivity. The method involves monitoring a handover condition, such as signal strength or network load, to determine when a handover is necessary. When the condition is met, the first network device initiates the handover process by sending a handover command to the mobile device. Additionally, the method includes transmitting data related to the handover condition being satisfied to the mobile device, providing it with context about the handover decision. This ensures the mobile device can prepare for the transition, improving handover reliability and reducing service interruptions. The method may also involve coordinating with other network devices to optimize the handover process, such as reserving resources in the target network before the handover occurs. By proactively managing handover conditions and communicating relevant data, the method enhances seamless connectivity for mobile users.
3. The method of claim 1 , further comprising: storing, by the first network device, the voice packet data to fill a capacity of the increased memory size of the dejitter buffer.
This invention relates to network communication systems, specifically methods for managing voice packet data in a dejitter buffer to improve real-time voice transmission quality. The problem addressed is the inefficiency in handling variable network delays, which can cause packet loss or jitter, degrading voice call quality. The solution involves dynamically adjusting the memory size of a dejitter buffer in a network device to accommodate fluctuations in packet arrival times, ensuring smoother playback. The method includes monitoring the arrival times of voice packets at a first network device, such as a router or switch, and calculating a delay variation metric. Based on this metric, the device adjusts the memory size of the dejitter buffer to compensate for network delays. If the delay variation exceeds a threshold, the buffer size is increased to store additional packets, preventing loss due to late arrivals. Conversely, if delays are minimal, the buffer size is reduced to minimize latency. The system also tracks packet sequence numbers to detect missing or out-of-order packets, further optimizing buffer management. Additionally, the method stores voice packet data in the dejitter buffer to fully utilize the increased memory capacity when necessary, ensuring efficient use of available resources. This dynamic adjustment process continuously adapts to network conditions, maintaining high-quality voice transmission even in unstable network environments. The invention is particularly useful in VoIP (Voice over IP) systems where consistent packet delivery is critical.
4. The method of claim 1 , wherein the punctuation type is a comma type.
A system and method for processing text data involves analyzing input text to identify and classify punctuation marks, particularly focusing on comma types. The method includes receiving a text input containing punctuation marks, detecting the presence of punctuation, and determining the specific type of punctuation, such as commas. The system then processes the identified punctuation to enhance text readability, formatting, or analysis. The comma type classification may involve distinguishing between different comma usage contexts, such as separating items in a list, setting off clauses, or indicating pauses. The method may further include applying rules or algorithms to modify, replace, or remove the identified comma types based on predefined criteria. The system may also integrate with text editing tools, natural language processing (NLP) applications, or document formatting software to automate punctuation handling. The goal is to improve text clarity, consistency, and machine readability by accurately identifying and managing comma types in various text processing tasks.
5. The method of claim 1 , further comprising: generating, by the first network device, text data based on the voice packet data, wherein the text data comprises punctuation data representative of a punctuation.
This invention relates to voice communication systems and addresses the challenge of accurately converting spoken language into text while preserving punctuation. The method involves a first network device processing voice packet data, which contains digitized speech, to generate corresponding text data. The text data includes punctuation data that represents punctuation marks such as periods, commas, or question marks, ensuring the transcribed text maintains proper grammatical structure and readability. The system may also involve a second network device that receives and processes the voice packet data, potentially performing additional tasks such as noise reduction or speech recognition. The first network device may further analyze the voice packet data to identify pauses, tone changes, or other speech patterns that indicate where punctuation should be inserted. The generated text data, including punctuation, can then be transmitted to another device or system for further use, such as real-time captioning, transcription services, or voice-controlled applications. This method improves the accuracy and usability of voice-to-text systems by ensuring the output text is properly formatted and grammatically correct.
6. The method of claim 5 , wherein the generating being performed is an indication of the voice activity having occurred.
This invention relates to voice activity detection (VAD) systems, which identify periods of speech in an audio signal. The problem addressed is the need for accurate and efficient detection of voice activity in noisy environments, where distinguishing speech from background noise is challenging. The invention improves upon prior methods by generating a clear indication of whether voice activity has occurred, enhancing reliability in applications such as speech recognition, telecommunication, and voice-controlled systems. The method involves analyzing an audio signal to determine the presence of speech. It includes processing the signal to extract features that distinguish speech from non-speech sounds, such as noise or silence. The system then evaluates these features to generate a binary or probabilistic output indicating whether voice activity was detected. This output can be used to trigger further actions, such as activating a speech recognition module or adjusting audio processing parameters. The invention may also incorporate adaptive thresholds or machine learning models to improve detection accuracy in varying acoustic conditions. By providing a clear indication of voice activity, the method ensures that downstream systems receive reliable input, reducing false activations and improving overall performance. The approach is particularly useful in real-time applications where quick and accurate detection is critical.
7. The method of claim 1 , further comprising: based on the punctuation type, predicting, by the first network device, the gap in the voice packet data.
This invention relates to voice communication systems, specifically improving the handling of voice packet data to reduce gaps in speech transmission. The problem addressed is the disruption in voice communication caused by gaps in transmitted voice packet data, which can occur due to network delays, packet loss, or improper punctuation handling. The solution involves a system where a first network device processes voice packet data and predicts gaps based on punctuation types detected in the data. The prediction is used to mitigate or fill these gaps, ensuring smoother and more continuous voice transmission. The method includes analyzing the voice packet data to identify punctuation marks, such as commas, periods, or question marks, and using these punctuation types to predict where gaps in the voice data may occur. The prediction is then used to adjust the transmission or playback of the voice data, such as by inserting silence or adjusting timing to compensate for the predicted gaps. This approach enhances the quality of voice communication by reducing perceived interruptions, particularly in real-time applications like VoIP (Voice over IP) or teleconferencing systems. The system may also involve additional processing steps, such as buffering or prioritizing certain packets, to further improve the handling of voice data. The invention aims to provide a more seamless and natural voice communication experience by intelligently predicting and addressing gaps in transmitted voice data.
8. A system, comprising: a processor; and a memory that stores executable instructions that, when executed by the processor, facilitate performance of operations, comprising: receiving voice packet data associated with a signal transfer of a mobile device from being connected to a first network device to being connected to a second network device, wherein the voice packet data comprises data related to a voice inactivity, and wherein the signal transfer is to occur at a first time; analyzing the data related to the voice inactivity to determine a punctuation type; in response to the punctuation type being determined by the analyzing, deferring the signal transfer based on the data related to the voice inactivity, resulting in a deferred signal transfer to occur at a second time after the first time; and converting the punctuation type to text data, wherein the text data comprises punctuation data representative of the punctuation.
This invention relates to a system for improving voice communication during network handoffs in mobile devices. The problem addressed is the disruption of voice calls when a mobile device transitions between network devices, particularly during periods of voice inactivity. The system includes a processor and memory storing executable instructions that perform operations to enhance call continuity. The system receives voice packet data associated with a signal transfer of a mobile device from a first network device to a second network device. The voice packet data includes information about voice inactivity, and the signal transfer is initially scheduled to occur at a first time. The system analyzes the voice inactivity data to determine the type of punctuation (e.g., comma, period, question mark) being used in the conversation. Based on this analysis, the system defers the signal transfer to a second time after the first time, ensuring the handoff occurs at a more appropriate moment, such as during a natural pause in speech. Additionally, the system converts the determined punctuation type into text data, which includes punctuation data representing the punctuation in written form. This allows for seamless integration of voice-to-text features while maintaining call quality during network transitions. The invention improves user experience by reducing call interruptions and enhancing the accuracy of voice-to-text conversions.
9. The system of claim 8 , wherein the operations further comprise: generating an indication indicative of a transfer condition being determined to have been satisfied.
A system for monitoring and managing data transfers in a computing environment addresses the problem of ensuring reliable and secure data movement between systems. The system includes a data transfer module that facilitates the movement of data between a source and a destination, a monitoring module that tracks the progress and status of the transfer, and a validation module that verifies the integrity and security of the transferred data. The system also includes a condition detection module that evaluates whether predefined transfer conditions, such as completion, failure, or security breaches, have been met. When a transfer condition is detected, the system generates an indication, such as an alert, log entry, or notification, to inform relevant stakeholders or trigger further actions. This indication ensures that appropriate responses, such as retries, security measures, or administrative actions, can be taken promptly. The system may also include a reporting module to document transfer events and conditions for auditing and compliance purposes. By providing real-time monitoring and automated condition detection, the system enhances data transfer reliability, security, and accountability in distributed computing environments.
10. The system of claim 8 , wherein the operations further comprise: determining a transfer interruption length, resulting in transfer interruption length data representative of the transfer interruption length.
A system for managing data transfers in a networked environment addresses the problem of optimizing transfer efficiency by minimizing interruptions during data transmission. The system monitors data transfer operations to detect interruptions, such as delays or disruptions, and analyzes these interruptions to determine their duration. The system generates transfer interruption length data, which quantifies the length of each detected interruption. This data can be used to assess transfer reliability, identify recurring issues, and implement corrective measures to improve transfer performance. The system may also include components for initiating data transfers, monitoring transfer progress, and storing transfer-related data. By quantifying interruption lengths, the system enables more precise diagnostics and proactive management of data transfer processes, enhancing overall network efficiency and reliability. The solution is particularly useful in environments where uninterrupted data flow is critical, such as cloud computing, real-time data processing, or high-speed network communications. The system's ability to measure and analyze transfer interruptions provides actionable insights for optimizing transfer protocols and reducing downtime.
11. The system of claim 8 , wherein the converting is performed in response to the voice inactivity being determined to have occurred.
A system for processing audio signals includes a voice activity detection module that monitors an audio input to detect periods of voice inactivity. When voice inactivity is detected, the system converts the audio signal from a first format to a second format. The first format may be a high-quality, high-bitrate format suitable for active voice transmission, while the second format may be a lower-quality, low-bitrate format to conserve bandwidth during periods of silence or inactivity. The system also includes a buffer to temporarily store the audio signal before conversion and a transmission module to send the converted signal to a remote device. The voice activity detection module may use threshold-based or machine learning techniques to distinguish between active speech and background noise. The conversion process may involve downsampling, bitrate reduction, or encoding the signal into a compressed format. The system ensures efficient use of network resources by dynamically adjusting the audio format based on real-time voice activity detection. This approach is particularly useful in communication applications where bandwidth optimization is critical, such as video conferencing, VoIP, or real-time collaboration tools. The system may also include error correction mechanisms to maintain audio quality during transmission.
12. The system of claim 8 , wherein the operations further comprise: generating an indication that the second network device is ready for the signal transfer.
A system for network device communication involves managing signal transfers between devices. The system includes a first network device and a second network device, where the first device is configured to receive a signal from an external source and the second device is configured to process the signal. The system also includes a controller that monitors the operational state of the second device to determine its readiness to receive the signal. The controller generates an indication when the second device is ready for the signal transfer, ensuring efficient and timely data processing. This readiness indication may involve checking the second device's power state, processing capacity, or other operational parameters to confirm it can handle the incoming signal without errors or delays. The system may also include mechanisms to delay or adjust the signal transfer if the second device is not ready, preventing data loss or corruption. The overall goal is to optimize signal transfer reliability and performance in networked environments.
13. The system of claim 8 , wherein the operations further comprise: receiving distance data in relation to respective distances of the mobile device to the first network device and the second network device.
A system for mobile device positioning and network connectivity management involves determining the location of a mobile device relative to multiple network devices. The system addresses the challenge of accurately positioning a mobile device in environments where traditional GPS signals may be weak or unavailable, such as indoors or in urban canyons. The system includes a mobile device and at least two network devices, each capable of communicating with the mobile device. The mobile device measures and receives distance data indicating its proximity to each network device. This distance data is used to calculate the mobile device's position, improving location accuracy and enabling applications like indoor navigation, asset tracking, and emergency services. The system may also optimize network connectivity by selecting the most efficient network device for communication based on the distance data. The mobile device may further adjust its transmission power or communication parameters to conserve energy and reduce interference. The system enhances both location accuracy and network efficiency in challenging environments.
14. The system of claim 8 , wherein the operations further comprise: in response to the converting the punctuation type to the text data, initiating the signal transfer between the first network device and the second network device.
This invention relates to a network communication system that converts punctuation types into text data to facilitate signal transfer between network devices. The system addresses the challenge of efficiently managing and transmitting punctuation marks in digital communication, ensuring accurate data interpretation and seamless signal transfer. The system includes a first network device and a second network device connected via a communication network. The first network device receives input data containing punctuation marks, such as commas, periods, or question marks. A processing module within the system converts these punctuation marks into corresponding text data, such as ASCII or Unicode representations. This conversion ensures compatibility across different network protocols and devices. Upon converting the punctuation type to text data, the system initiates a signal transfer between the first and second network devices. The signal transfer may involve transmitting the converted text data over a wired or wireless network, ensuring reliable communication. The system may also include error-checking mechanisms to verify the integrity of the transmitted data. The invention improves network communication by standardizing punctuation handling, reducing transmission errors, and enhancing interoperability between diverse network devices. This ensures accurate data exchange in applications such as messaging, document sharing, or real-time collaboration.
15. A non-transitory machine-readable medium, comprising executable instructions that, when executed by a processor, facilitate performance of operations, comprising: receiving voice packet data related to a network device transfer detection message associated with a signal of a mobile device; based on a determination, in response to the receiving, that the voice packet data comprises dormancy packet data related to the voice packet data, increasing a dejitter buffer from a first capacity to a second capacity, wherein the dejitter buffer decreases a voice packet delay variation associated with queuing the voice packet data as compared to without the dejitter buffer; and in response to the determination, converting the voice packet data to text data, wherein the text data comprises punctuation data representative of a punctuation type.
This invention relates to improving voice packet processing in mobile networks, particularly for handling dormancy states and reducing delay variations. The system receives voice packet data from a mobile device, which may include network transfer detection messages. When the voice packet data indicates dormancy (a low-activity state), the system dynamically increases the capacity of a dejitter buffer. The dejitter buffer mitigates voice packet delay variations by smoothing out timing inconsistencies in the queued voice packets. After adjusting the buffer, the system converts the voice packets into text data, including punctuation marks to enhance readability. This approach ensures smoother voice transmission during dormancy periods while enabling accurate text conversion for applications like transcription or voice-to-text services. The dynamic buffer adjustment optimizes performance by adapting to network conditions, reducing latency, and improving voice quality in mobile communications.
16. The non-transitory machine-readable medium of claim 15 , wherein the determination is a first determination, and wherein the operations further comprise: based on a second determination of a punctuation type from the dormancy packet data, deferring a signal network device transfer based on the dormancy packet data, resulting in a deferred signal network device transfer.
This invention relates to network communication systems, specifically methods for managing signal network device transfers in data transmission. The problem addressed is the inefficient handling of dormancy packet data, which can lead to unnecessary or premature transfers of network devices, causing delays and resource waste. The invention involves a non-transitory machine-readable medium storing instructions that, when executed, perform operations to optimize network device transfers. A first determination is made regarding the content of dormancy packet data, which may include identifying specific data types or patterns. Based on a second determination of the punctuation type within the dormancy packet data, the system defers a signal network device transfer. This deferral is conditional on the dormancy packet data, ensuring that transfers only occur when necessary, thereby improving efficiency and reducing unnecessary resource consumption. The deferred transfer results in a more optimized network operation, minimizing disruptions and conserving energy or bandwidth. The system dynamically adjusts transfer decisions based on real-time analysis of packet data, enhancing overall network performance.
17. The non-transitory machine-readable medium of claim 15 , wherein the operations further comprise: storing the voice packet data in the dejitter buffer until the dejitter buffer is determined to be at the second capacity.
This invention relates to network communication systems, specifically methods for managing voice packet data in a dejitter buffer to improve real-time audio transmission quality. The problem addressed is the variability in packet arrival times in network communications, which can cause delays, jitter, and audio quality degradation. The solution involves dynamically adjusting the capacity of a dejitter buffer to optimize playback timing and reduce latency. The system processes incoming voice packet data by storing it in a dejitter buffer, which temporarily holds packets to compensate for network jitter. The buffer's capacity is dynamically adjusted based on network conditions. When the buffer reaches a first capacity threshold, it begins releasing packets for playback at a controlled rate. If the buffer reaches a second, higher capacity threshold, additional packets are stored until the buffer is full, ensuring smooth playback without excessive delay. The system monitors packet arrival times and adjusts the buffer capacity in real-time to maintain optimal performance. This approach improves audio quality by reducing gaps or overlaps in playback while minimizing latency. The dynamic adjustment of buffer capacity ensures that the system adapts to varying network conditions, providing a more reliable and consistent audio experience. The invention is particularly useful in voice-over-IP (VoIP) and real-time communication applications where minimizing delay and maintaining audio clarity are critical.
18. The non-transitory machine-readable medium of claim 15 , wherein the operations further comprise: sending an indication that the dejitter buffer is at the second capacity to the mobile device.
A system for managing a dejitter buffer in a communication network addresses the problem of packet delay variation (jitter) in real-time communication, which can degrade audio or video quality. The dejitter buffer temporarily stores incoming packets to smooth out timing irregularities, but its capacity must be dynamically adjusted to balance latency and packet loss. The invention provides a method to monitor the buffer's fill level and adjust its capacity based on network conditions. When the buffer reaches a first threshold, its capacity is increased to reduce packet loss risk. If the buffer reaches a second, higher threshold, indicating excessive latency, its capacity is decreased to minimize delay. The system also sends an indication to the mobile device when the buffer is at the second capacity, allowing the device to adjust its transmission rate or buffer management accordingly. This dynamic adjustment improves real-time communication quality by adapting to varying network conditions while maintaining low latency and minimizing packet loss.
19. The non-transitory machine-readable medium of claim 15 , wherein the operations further comprise: associating punctuation data with the dormancy packet data, and wherein the punctuation data comprises the punctuation type.
A system and method for network communication involves transmitting and receiving data packets, including dormancy packets, to manage device power states. The invention addresses the challenge of efficiently handling communication in low-power or dormant states while maintaining data integrity. The system generates and processes dormancy packets, which are specialized data packets used to signal or maintain a dormant state in a communication device. These packets may include metadata or control information to manage power states, synchronization, or other operational parameters. The invention further enhances dormancy packet functionality by associating punctuation data with the packet data. Punctuation data includes a punctuation type, which categorizes or defines the purpose of the punctuation within the packet. This punctuation type may indicate the role of the punctuation in the packet structure, such as marking boundaries, signaling transitions, or providing synchronization points. The punctuation data and its type help ensure proper interpretation and processing of the dormancy packet, improving communication reliability and efficiency in low-power states. The system may be implemented in various networked devices, including wireless communication systems, IoT devices, or other power-sensitive applications.
20. The non-transitory machine-readable medium of claim 15 , wherein the increasing the dejitter buffer is proportional to a value associated with the punctuation type.
This invention relates to audio processing, specifically to adjusting a dejitter buffer in real-time communication systems to improve speech clarity. The problem addressed is the disruption caused by abrupt pauses or delays when punctuation marks (e.g., commas, periods) are inserted into speech streams, which can degrade user experience. The solution involves dynamically increasing the size of a dejitter buffer based on the type of punctuation detected in the audio stream. The buffer adjustment is proportional to a predefined value associated with each punctuation type, ensuring smoother transitions and reducing perceived interruptions. The system first analyzes the audio stream to identify punctuation marks, then applies a corresponding buffer adjustment factor to the dejitter buffer. Different punctuation types (e.g., commas, periods, question marks) may have different adjustment values to reflect their typical impact on speech flow. The buffer size is increased proportionally to the punctuation type's value, allowing the system to accommodate the natural pauses associated with punctuation without introducing artificial delays. This approach enhances speech intelligibility and user experience in real-time communication applications.
Unknown
August 11, 2020
Browse 5M+ US patents with plain-English claim translations and AI-generated analysis.